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@ -295,8 +295,8 @@ typedef struct tagBITMAPINFOHEADER { |
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#define MAX_AUDIO_BUFFER_POOL_CHANNELS 16 // Audio pool channels |
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#endif |
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#ifndef AUDIO_BUFFER_CONVERSION_CACHE_SIZE |
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#define AUDIO_BUFFER_CONVERSION_CACHE_SIZE 256 // In PCM frames. Smaller values use less memory but have more overhead.. |
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#ifndef AUDIO_BUFFER_RESIDUAL_CAPACITY |
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#define AUDIO_BUFFER_RESIDUAL_CAPACITY 8 // In PCM frames. For resampling and pitch shifting. |
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#endif |
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//---------------------------------------------------------------------------------- |
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@ -341,9 +341,8 @@ typedef enum { |
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// Audio buffer struct |
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struct rAudioBuffer { |
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ma_data_converter converter; // Audio data converter |
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unsigned char* converterCache; // Cached input samples for use by the converter when resampling is required |
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unsigned int converterCacheCap; // The capacity of the converter cache in frames |
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unsigned int converterCacheLen; // The number of valid frames sitting in the converter cache |
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unsigned char* converterResidual; // Cached residual input frames for use by the converter |
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unsigned int converterResidualCount; // The number of valid frames sitting in converterResidual |
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AudioCallback callback; // Audio buffer callback for buffer filling on audio threads |
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rAudioProcessor *processor; // Audio processor |
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@ -599,9 +598,8 @@ AudioBuffer *LoadAudioBuffer(ma_format format, ma_uint32 channels, ma_uint32 sam |
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// ensure there are no discontinuities. Since raylib supports pitch |
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// shifting, which is done through resampling, a cache will always be |
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// required. This will be kept relatively small to avoid too much wastage. |
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audioBuffer->converterCacheLen = 0; |
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audioBuffer->converterCacheCap = AUDIO_BUFFER_CONVERSION_CACHE_SIZE; |
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audioBuffer->converterCache = (unsigned char*)RL_CALLOC(audioBuffer->converterCacheCap*ma_get_bytes_per_frame(format, channels), 1); |
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audioBuffer->converterResidualCount = 0; |
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audioBuffer->converterResidual = (unsigned char*)RL_CALLOC(AUDIO_BUFFER_RESIDUAL_CAPACITY*ma_get_bytes_per_frame(format, channels), 1); |
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// Init audio buffer values |
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audioBuffer->volume = 1.0f; |
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@ -638,7 +636,7 @@ void UnloadAudioBuffer(AudioBuffer *buffer) |
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{ |
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UntrackAudioBuffer(buffer); |
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ma_data_converter_uninit(&buffer->converter, NULL); |
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RL_FREE(buffer->converterCache); |
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RL_FREE(buffer->converterResidual); |
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RL_FREE(buffer->data); |
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RL_FREE(buffer); |
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} |
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@ -2474,47 +2472,77 @@ static ma_uint32 ReadAudioBufferFramesInMixingFormat(AudioBuffer *audioBuffer, f |
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// NOTE: Continuously converting data from the AudioBuffer's internal format to the mixing format, |
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// which should be defined by the output format of the data converter. |
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// This is done until frameCount frames have been output. |
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// A cache is required to ensure continuity when resampling. |
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ma_uint32 bpf = ma_get_bytes_per_frame(audioBuffer->converter.formatIn, audioBuffer->converter.channelsIn); |
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ma_uint8 inputBuffer[4096] = { 0 }; |
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ma_uint32 inputBufferFrameCap = sizeof(inputBuffer)/bpf; |
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ma_uint32 totalOutputFramesProcessed = 0; |
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while (totalOutputFramesProcessed < frameCount) |
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{ |
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float *runningFramesOut = framesOut + (totalOutputFramesProcessed*audioBuffer->converter.channelsOut); |
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ma_uint64 outputFramesToProcessThisIteration = frameCount - totalOutputFramesProcessed; |
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ma_uint64 inputFramesToProcessThisIteration = 0; |
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// Output frames come from the converter. The converter reads from the cache. The process |
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// goes like this: |
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// |
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// AudioBuffer -> Cache -> Converter -> framesOut |
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// |
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// Data is moved from the AudioBuffer into the cache, and then the cache is fed into the |
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// converter which outputs to the output buffer. |
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// Refill the cache if necessary. |
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if (audioBuffer->converterCacheLen == 0) |
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// Process any residual input frames from the previous read first. |
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if (audioBuffer->converterResidualCount > 0) |
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{ |
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audioBuffer->converterCacheLen = ReadAudioBufferFramesInInternalFormat(audioBuffer, audioBuffer->converterCache, audioBuffer->converterCacheCap); |
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} |
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// Now run the data through the data converter. |
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if (audioBuffer->converterCacheLen > 0) |
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{ |
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ma_uint32 bpf = ma_get_bytes_per_frame(audioBuffer->converter.formatIn, audioBuffer->converter.channelsIn); |
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float *runningFramesOut = framesOut + (totalOutputFramesProcessed*audioBuffer->converter.channelsOut); |
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ma_uint64 inputFramesProcessedThisIteration = audioBuffer->converterCacheLen; |
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ma_uint64 inputFramesProcessedThisIteration = audioBuffer->converterResidualCount; |
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ma_uint64 outputFramesProcessedThisIteration = outputFramesToProcessThisIteration; |
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ma_data_converter_process_pcm_frames(&audioBuffer->converter, audioBuffer->converterCache, &inputFramesProcessedThisIteration, runningFramesOut, &outputFramesProcessedThisIteration); |
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ma_data_converter_process_pcm_frames(&audioBuffer->converter, audioBuffer->converterResidual, &inputFramesProcessedThisIteration, runningFramesOut, &outputFramesProcessedThisIteration); |
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// Make sure the data in the cache is consumed. This can be optimized to use a cursor instead of a memmove(). |
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memmove(audioBuffer->converterCache, audioBuffer->converterCache + inputFramesProcessedThisIteration*bpf, (size_t)(audioBuffer->converterCacheCap - inputFramesProcessedThisIteration) * bpf); |
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audioBuffer->converterCacheLen -= (ma_uint32)inputFramesProcessedThisIteration; // Safe cast |
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memmove(audioBuffer->converterResidual, audioBuffer->converterResidual + inputFramesProcessedThisIteration*bpf, (size_t)(AUDIO_BUFFER_RESIDUAL_CAPACITY - inputFramesProcessedThisIteration) * bpf); |
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audioBuffer->converterResidualCount -= (ma_uint32)inputFramesProcessedThisIteration; // Safe cast |
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totalOutputFramesProcessed += (ma_uint32)outputFramesProcessedThisIteration; // Safe cast |
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} |
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else |
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{ |
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break; // Ran out of input data. |
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// Getting here means there are no residual frames from the previous read. Fresh data can now be |
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// pulled from the AudioBuffer and processed. |
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// |
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// A best guess needs to be used made to determine how many input frames to pull from the |
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// buffer. There are three possible outcomes: 1) exact; 2) underestimated; 3) overestimated. |
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// |
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// When the guess is exactly correct or underestimated there is nothing special to handle - it'll be |
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// handled naturally by the loop. |
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// |
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// When the guess is overestimated, that's when it gets more complicated. In this case, any overflow |
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// needs to be stored in a buffer for later processing by the next read. |
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ma_uint32 estimatedInputFrameCount = (ma_uint32)(((float)audioBuffer->converter.resampler.sampleRateIn / audioBuffer->converter.resampler.sampleRateOut) * outputFramesToProcessThisIteration); |
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if (estimatedInputFrameCount == 0) |
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{ |
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estimatedInputFrameCount = 1; // Make sure at least one input frame is read. |
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} |
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if (estimatedInputFrameCount > inputBufferFrameCap) |
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{ |
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estimatedInputFrameCount = inputBufferFrameCap; |
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} |
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estimatedInputFrameCount = ReadAudioBufferFramesInInternalFormat(audioBuffer, inputBuffer, estimatedInputFrameCount); |
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ma_uint64 inputFramesProcessedThisIteration = estimatedInputFrameCount; |
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ma_uint64 outputFramesProcessedThisIteration = outputFramesToProcessThisIteration; |
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ma_data_converter_process_pcm_frames(&audioBuffer->converter, inputBuffer, &inputFramesProcessedThisIteration, runningFramesOut, &outputFramesProcessedThisIteration); |
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if (estimatedInputFrameCount > inputFramesProcessedThisIteration) |
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{ |
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// Getting here means the estimated input frame count was overestimated. The residual needs |
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// be stored for later use. |
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ma_uint64 residualFrameCount = estimatedInputFrameCount - inputFramesProcessedThisIteration; |
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// A safety check to make sure the capacity of the residual cache is not exceeded. |
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if (residualFrameCount > AUDIO_BUFFER_RESIDUAL_CAPACITY) |
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{ |
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residualFrameCount = AUDIO_BUFFER_RESIDUAL_CAPACITY; |
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} |
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memcpy(audioBuffer->converterResidual, inputBuffer + inputFramesProcessedThisIteration*bpf, (size_t)(residualFrameCount * bpf)); |
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audioBuffer->converterResidualCount = residualFrameCount; |
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} |
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totalOutputFramesProcessed += (ma_uint32)outputFramesProcessedThisIteration; |
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} |
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} |
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